1. Technical Field
The present invention relates generally to an audio playback apparatus for decoding and reproducing an audio signal encoded in frames, and relates more specifically to a playback apparatus and playback method for reproducing audio without producing noise when attributes change or there is a data discontinuity in the audio signal due to editing or a communication error.
2. Background Art
Playback methods for decoding and presenting audio signals encoded as digital code streams are widely available today in the form of playback devices and computer programs for listening to music and other audio content. In most such implementations the audio signal is encoded in audio data frames according to the MPEG standard, particularly ISO 11172-3 or ISO 13818-3. A private header containing signal attributes is added to each frame. A CRC bit for error checking is also added to the encoded audio signal, thus enabling checking during the decoding process for data errors and data loss on the transmission path.
However, when data loss on the transmission path is high, resulting in discontinuities in the data stream, error correction cannot restore the signal. Outputting the audio signal with such data discontinuities produces noise. To eliminate this noise, the audio is preferably muted.
An example of a conventional playback apparatus is taught in Japanese Unexamined Patent Application Publication 2000-259195. Instead of detecting these signal discontinuities, this playback apparatus detects changes in settings from the transmission side, such as changes in the sampling frequency in the data stream, and mutes audio output for a predetermined time after such a change is detected. When there is such a change, the receiver must automatically adjust to the changed setting, and mutes the audio output so that noise is not produced during the automatic adjustment. This conventional playback apparatus detects a valid header and compares the sampling frequency written to the one previous valid header interpreted by a header interpreting means with the sampling frequency written in the current valid header currently being decoded. If the sampling frequency in the current header has changed, audio is muted for a specific time in the frame following the sampling frequency change to prevent outputting noise.
If the sampling frequency written in the current header is different from the sampling frequency in the preceding header, for example, the operating parameters of the DA converter downstream from the decoding means must be changed. Furthermore, because a correct audio signal will not be produced while the DA converter settings are being changed, the output audio signal will contain noise. As a result, audio output is muted for the time required to change the DA converter settings. Audio is therefore muted for the frame containing the header with the changed setting and one or more subsequent frames.
The header is detected by detecting a synchronization word (“syncword”), which is set and used for synchronization with the header.
This syncword is further described in Japanese Unexamined Patent Application Publication 2000-31942.
Japanese Unexamined Patent Application Publication H10-209876 teaches a muting process that detects lost data by comparing the data size to apply muting. The conventional bitstream playback apparatus taught in Japanese Unexamined Patent Application Publication H10-209876 decodes an audio stream encoded to the MPEG-1 or MPEG-2 Audio standard, detects a frame buffer underflow in the decoder when part of the bitstream is lost for any reason, and thus mutes output. More specifically, this apparatus detects the syncword to find valid headers, and counts the data between one valid header and another valid header. If the counted data size F is less than a predetermined size, data loss is detected and muting is applied.